From: Olivier Crête Date: Wed, 31 Mar 2021 15:41:45 +0000 (-0400) Subject: webrtcbin test: Add for the case where a second m-line is renegotiated X-Git-Tag: 1.19.3~507^2~528 X-Git-Url: http://review.tizen.org/git/?a=commitdiff_plain;h=4a48e291ffae9c73c0c0e7507e18a64652abbc24;p=platform%2Fupstream%2Fgstreamer.git webrtcbin test: Add for the case where a second m-line is renegotiated This is for the case where there answerer forces a specific media type for a m-line, but he origin offer only has the other media type. In this case, we will create a second transceiver on receiving the offer and add the desired media type using renegotiation. Part-of: --- diff --git a/tests/check/elements/webrtcbin.c b/tests/check/elements/webrtcbin.c index 3618627..d3c8dc7 100644 --- a/tests/check/elements/webrtcbin.c +++ b/tests/check/elements/webrtcbin.c @@ -37,7 +37,7 @@ #define OPUS_RTP_CAPS(pt) "application/x-rtp,payload=" G_STRINGIFY(pt) ",encoding-name=OPUS,media=audio,clock-rate=48000,ssrc=(uint)3384078950" #define VP8_RTP_CAPS(pt) "application/x-rtp,payload=" G_STRINGIFY(pt) ",encoding-name=VP8,media=video,clock-rate=90000,ssrc=(uint)3484078950" -#define TEST_IS_OFFER_ELEMENT(t, e) ((t)->offerror == 1 && (e) == (t)->webrtc1 ? TRUE : FALSE) +#define TEST_IS_OFFER_ELEMENT(t, e) ((((t)->offerror == 1 && (e) == (t)->webrtc1) || ((t)->offerror == 2 && (e) == (t)->webrtc2)) ? TRUE : FALSE) #define TEST_GET_OFFEROR(t) (TEST_IS_OFFER_ELEMENT(t, t->webrtc1) ? (t)->webrtc1 : t->webrtc2) #define TEST_GET_ANSWERER(t) (TEST_IS_OFFER_ELEMENT(t, t->webrtc1) ? (t)->webrtc2 : t->webrtc1) @@ -531,6 +531,7 @@ test_webrtc_new (void) ret->on_answer_created = _offer_answer_not_reached; ret->on_data_channel = _on_data_channel_not_reached; ret->bus_message = _bus_no_errors; + ret->offerror = 1; g_mutex_init (&ret->lock); g_cond_init (&ret->cond); @@ -661,13 +662,13 @@ test_webrtc_free (struct test_webrtc *t) } static void -test_webrtc_create_offer (struct test_webrtc *t, GstElement * webrtc) +test_webrtc_create_offer (struct test_webrtc *t) { GstPromise *promise; + GstElement *offeror = TEST_GET_OFFEROR (t); - t->offerror = webrtc == t->webrtc1 ? 1 : 2; promise = gst_promise_new_with_change_func (_on_offer_received, t, NULL); - g_signal_emit_by_name (webrtc, "create-offer", NULL, promise); + g_signal_emit_by_name (offeror, "create-offer", NULL, promise); } static void @@ -748,7 +749,7 @@ on_negotiation_needed_hit (struct test_webrtc *t, GstElement * element, { guint *flag = (guint *) user_data; - *flag = 1; + *flag |= 1 << ((element == t->webrtc1) ? 1 : 2); } typedef void (*ValidateSDPFunc) (struct test_webrtc * t, GstElement * element, @@ -772,7 +773,7 @@ _check_validate_sdp (struct test_webrtc *t, GstElement * element, struct validate_sdp *validate = user_data; GstWebRTCSessionDescription *desc = NULL; - if (t->offerror == 1 && t->webrtc1 == element) + if (TEST_IS_OFFER_ELEMENT (t, element)) desc = t->offer_desc; else desc = t->answer_desc; @@ -810,7 +811,7 @@ test_validate_sdp_full (struct test_webrtc *t, struct validate_sdp *offer, GST_STATE_READY) == GST_STATE_CHANGE_FAILURE); } - test_webrtc_create_offer (t, t->webrtc1); + test_webrtc_create_offer (t); if (wait_mask == 0) { test_webrtc_wait_for_answer_error_eos (t); @@ -2564,7 +2565,7 @@ GST_START_TEST (test_duplicate_nego) t->harnesses = g_list_prepend (t->harnesses, h); test_validate_sdp (t, &offer, &answer); - fail_unless_equals_int (negotiation_flag, 1); + fail_unless (negotiation_flag & (1 << 2)); test_webrtc_reset_negotiation (t); test_validate_sdp (t, &offer, &answer); @@ -3297,7 +3298,7 @@ GST_START_TEST (test_renego_lose_media_fails) t->on_offer_set = offer_set_produced_error; t->on_answer_created = NULL; - test_webrtc_create_offer (t, t->webrtc1); + test_webrtc_create_offer (t); test_webrtc_wait_for_state_mask (t, 1 << STATE_CUSTOM); test_webrtc_free (t); @@ -3603,6 +3604,97 @@ GST_START_TEST (test_reject_set_description) GST_END_TEST; +GST_START_TEST (test_force_second_media) +{ + struct test_webrtc *t = test_webrtc_new (); + const gchar *media_types[] = { "audio" }; + VAL_SDP_INIT (media_type, _verify_media_types, &media_types, NULL); + guint media_format_count[] = { 1, }; + VAL_SDP_INIT (media_formats, on_sdp_media_count_formats, + media_format_count, &media_type); + const gchar *expected_offer_setup[] = { "actpass", }; + VAL_SDP_INIT (offer_setup, on_sdp_media_setup, expected_offer_setup, + &media_formats); + const gchar *expected_answer_setup[] = { "active", }; + VAL_SDP_INIT (answer_setup, on_sdp_media_setup, expected_answer_setup, + &media_formats); + const gchar *expected_offer_direction[] = { "sendrecv", }; + VAL_SDP_INIT (offer_direction, on_sdp_media_direction, + expected_offer_direction, &offer_setup); + const gchar *expected_answer_direction[] = { "recvonly", }; + VAL_SDP_INIT (answer_direction, on_sdp_media_direction, + expected_answer_direction, &answer_setup); + VAL_SDP_INIT (answer_count, _count_num_sdp_media, GUINT_TO_POINTER (1), + &answer_direction); + VAL_SDP_INIT (offer_count, _count_num_sdp_media, GUINT_TO_POINTER (1), + &offer_direction); + + const gchar *second_media_types[] = { "audio", "video" }; + VAL_SDP_INIT (second_media_type, _verify_media_types, &second_media_types, + NULL); + guint second_media_format_count[] = { 1, 1 }; + VAL_SDP_INIT (second_media_formats, on_sdp_media_count_formats, + second_media_format_count, &second_media_type); + const gchar *second_expected_offer_setup[] = { "active", "actpass" }; + VAL_SDP_INIT (second_offer_setup, on_sdp_media_setup, + second_expected_offer_setup, &second_media_formats); + const gchar *second_expected_answer_setup[] = { "passive", "active" }; + VAL_SDP_INIT (second_answer_setup, on_sdp_media_setup, + second_expected_answer_setup, &second_media_formats); + const gchar *second_expected_answer_direction[] = { "sendonly", "recvonly" }; + VAL_SDP_INIT (second_answer_direction, on_sdp_media_direction, + second_expected_answer_direction, &second_answer_setup); + const gchar *second_expected_offer_direction[] = { "recvonly", "sendrecv" }; + VAL_SDP_INIT (second_offer_direction, on_sdp_media_direction, + second_expected_offer_direction, &second_offer_setup); + VAL_SDP_INIT (second_answer_count, _count_num_sdp_media, GUINT_TO_POINTER (2), + &second_answer_direction); + VAL_SDP_INIT (second_offer_count, _count_num_sdp_media, GUINT_TO_POINTER (2), + &second_offer_direction); + + GstHarness *h; + guint negotiation_flag = 0; + GstPadTemplate *templ; + GstCaps *caps; + GstPad *pad; + + /* add a transceiver that will only receive an opus stream and check that + * the created offer is marked as recvonly */ + t->on_negotiation_needed = on_negotiation_needed_hit; + t->negotiation_data = &negotiation_flag; + t->on_ice_candidate = NULL; + t->on_pad_added = _pad_added_fakesink; + + /* setup peer */ + h = gst_harness_new_with_element (t->webrtc1, "sink_0", NULL); + add_fake_audio_src_harness (h, 96); + t->harnesses = g_list_prepend (t->harnesses, h); + + /* Create a second side with specific video caps */ + templ = gst_element_get_pad_template (t->webrtc2, "sink_%u"); + fail_unless (templ != NULL); + caps = gst_caps_from_string (VP8_RTP_CAPS (97)); + pad = gst_element_request_pad (t->webrtc2, templ, NULL, caps); + gst_caps_unref (caps); + fail_unless (pad != NULL); + h = gst_harness_new_with_element (t->webrtc2, GST_PAD_NAME (pad), NULL); + gst_object_unref (pad); + add_fake_video_src_harness (h, 97); + t->harnesses = g_list_prepend (t->harnesses, h); + + test_validate_sdp (t, &offer_count, &answer_count); + fail_unless (negotiation_flag & 1 << 2); + + test_webrtc_reset_negotiation (t); + + t->offerror = 2; + test_validate_sdp (t, &second_offer_count, &second_answer_count); + + test_webrtc_free (t); +} + +GST_END_TEST; + static Suite * webrtcbin_suite (void) { @@ -3649,6 +3741,7 @@ webrtcbin_suite (void) tcase_add_test (tc, test_reject_request_pad); tcase_add_test (tc, test_reject_create_offer); tcase_add_test (tc, test_reject_set_description); + tcase_add_test (tc, test_force_second_media); if (sctpenc && sctpdec) { tcase_add_test (tc, test_data_channel_create); tcase_add_test (tc, test_data_channel_remote_notify);